[Asterisk-Users] Paging and Auto Answer on Grandstream GXP2000

Fri May 12 11:49:16 MST 2006

You are correct, That is why the PAGE() Application was made. It creates
a MeetMe room, calls the Technologies on the list and transfers the
calls to the temp MeetMe. 

________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Lacy Moore
- Aspendora
Sent: Friday, May 12, 2006 2:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Paging and Auto Answer on Grandstream
GXP2000

I must be missing something.  Seems to me that only one phone would
connect.  This is just a plain dial command that rings all those
extensions and when one answers, the rest stop ringing.

Right?

On 5/12/06, Tom Vile <tvile at baldwintechsolutions.com> wrote: 

make the dial command like so:

exten => PAGE203,n,Dial(SIP/203&SIP/204&SIP/205SIP/206,5) 

On 5/12/06, Forrest Beck <jonforrest.beck at gmail.com> wrote:
> Thanks.  I like that method.
>
> Do you think if I add all my extensions (say 40 of them) to new Dial 
> commands after "exten => PAGE203,n,Dial(SIP/203,5)"
>
> Like this:
> exten => PAGE203,n,Dial(SIP/203,5)
> exten => PAGE203,n,Dial(SIP/204,5)
> exten => PAGE203,n,Dial(SIP/205,5) 
> exten => PAGE203,n,Dial(SIP/206,5)
>
> will work?
>
> The only think I am not sure about is if a phone doesn't answer in the
5
> seconds allowed, it will hang until that 5 seconds is up to parse the
next 
> line.
>
> Also, if it is dialing each phone line by line will there be a delay
for all
> the phones are dialed and pick up.
>
> Thanks again.
>
> Forrest
>
> -----Original Message----- 
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com ] On Behalf Of Tom
Vile
> Sent: Friday, May 12, 2006 10:21 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Paging and Auto Answer on Grandstream
GXP2000 
>
> This is what I use:
>
> [ext-paging]
> exten => PAGE203,1,Set(__SIPADDHEADER=Call-Info: answer-after=0)
> exten => PAGE203,n,Set(__ALERT_INFO=Ring Answer)
> exten => PAGE203,n,Set(__SIP_URI_OPTIONS=intercom=true) 
> exten => PAGE203,n,Dial(SIP/203,5)
> exten => Debug,1,Noop(dialstr is LOCAL/PAGE203 at ext-paging)
> exten => 2030,1,Page(LOCAL/PAGE203 at ext-paging)
>
> if I dial 2030 it will page extension 203.  Change accordingly.  This 
> works on my Grandstream and Snom phones.
>
>
> On 5/12/06, Forrest Beck <jonforrest.beck at gmail.com> wrote:
> > Asterisk 1.2.7.1 and Zaptel 1.2.5
> >
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com
> > [mailto: asterisk-users-bounces at lists.digium.com] On Behalf Of Tom
Vile
> > Sent: Thursday, May 11, 2006 6:55 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion 
> > Subject: Re: [Asterisk-Users] Paging and Auto Answer on Grandstream
> GXP2000
> >
> > What version of Asterisk?
> >
> > On 5/11/06, Forrest Beck < jonforrest.beck at gmail.com
<mailto:jonforrest.beck at gmail.com> > wrote:
> > >
> > >
> > >
> > >
> > > I am looking to setup paging using the auto answer feature on the
> > > Grandstream GXP2000.  I am thinking I will follow the method as 
> described
> > > here:
> > >
> > >
> > >
> > > http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page 
> > >
> > >
> > >
> > > I will setup the 4th account on the phone to auto answer.
> > >
> > >
> > >
> > > Does anyone else have a method that works better?  I also looked
at the 
> > > allpage AGI written on Voip-Info.  But it seems to dial all
extensions,
> > even
> > > the ones I don't want to use for Auto Answer.
> > >
> > >
> > > 
> > > I really would like a way to group the phones instead of having
them all
> > > listed in a dial command.
> > >
> > >
> > >
> > > exten =>
> > >
7999,2,Page(Local/202 at page&Local/208 at page&Local/210 at page/n&Local/interal
> > > 223 at page|)
> > >
> > >
> > >
> > > Thanks!
> > > _______________________________________________ 
> > > --Bandwidth and Colocation provided by Easynews.com --
> > >
> > > Asterisk-Users mailing list
> > > To UNSUBSCRIBE or update options visit: 
> > >
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > >
> > >
> > 
> >
> > --
> > Tom Vile
> > Baldwin Technology Solutions, Inc
> > Consulting - Web Design - VoIP Telephony
> > www.baldwintechsolutions.com 
> > Phone: 518-631-2855 x205
> > Fax:     518-631-2856
> > _______________________________________________
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> >
> > Asterisk-Users mailing list
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> >
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> >
>
> 
> --
> Tom Vile
> Baldwin Technology Solutions, Inc
> Consulting - Web Design - VoIP Telephony
> www.baldwintechsolutions.com
> Phone: 518-631-2855 x205 
> Fax:     518-631-2856
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list 
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
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>

--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax:     518-631-2856
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-- 
Lacy Moore
Aspendora, Inc. 

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